Professional Audio

Professional audio, also 'pro audio', can be used a term to refer to both a type of audio equipment as well as a type of audio engineering application.

Professional audio equipment can be used to describe any audio equipment used or marketed for use as a sound application by or for a professional or professional purpose. This includes, but is not limited to, loudspeakers, microphones, Mixing consoles, amplifiers, recording and playback devices such as dat or turntables, and in some cases telephony devices. Pro Audio equipment typically carries an implied elevation of manufacturing quality and features compared to regular or consumer level audio equipment (as is common with other types of professional equipment.)

Professional audio application is commonly used to refer to professional audio engineering and operations, which can include but is not limited to broadcasting radio, audio mastering, sound reinforcement such as a concert, DJ performances, Audio Sampling , public address, surround sound movie theatres, and in some cases piped music application.

Both terms imply involvement of audio engineering at an industrial(occupational) level as opposed to a personal level. For example, a regular personal use microphone such as one in a mobile phone would have a very limited dynamic range focused on speech, whereas a pro audio microphone would have a much wider dynamic range to capture quiet whispers or loud musical instruments. A regular loudspeaker for home use may handle 100 watts rms at a given signal-to-noise ratio, whereas a pro audio loudspeaker such as one used for concert venues may handle 1000 watts rms or more, or a studio use speaker may operate at a significantly more efficient signal-to-noise rating at the same 100 watts as the home speaker.

Specifications alone do not inherently include or exclude equipment for consideration as professional audio level, but are used by most publications and documentation as a starting point of reference.

Soundproofing

Soundproofing is any means of reducing the intensity of sound with respect to a specified source and receptor. There are several basic approaches to reducing sound: increasing the distance between source and receiver, using noise barriers to block or absorb the energy of the sound waves, using damping structures such as sound baffles, or using active antinoise sound generators.

Soundproofing affects sound in two different ways: noise reduction and noise absorption. Noise reduction simply blocks the passage of sound waves through the use of distance and intervening objects in the sound path. Noise absorption, on the other hand, operates by transforming the sound wave. Noise absorption involves suppressing echoes, reverberation, resonance and reflection. The damping characteristics of the materials it is made out of are important in noise absorption.

Contents [hide]
1 Distance
2 Damping
3 Room Within A Room
4 Noise cancellation
5 Noise barriers as exterior soundproofing
6 See also
7 References
8 External link



[edit] Distance
The use of distance to dissipate sound is straightforward. The energy density of sound waves decrease as they spread out, so that increasing the distance between the receiver and source results in a progressively lesser intensity of sound at the receiver. In a normal three dimensional setting, the intensity of sound waves will be attenuated according to the inverse square of the distance from the source. Using mass to absorb sound is also quite straightforward, with part of the sound energy being used to vibrate the mass of the intervening object, rather than being transmitted. When this mass consists of air the extra dissipation on top of the distance effect is only significant for typically more than 1000 meters, depending also on the weather and reflections from the soil.[1]


[edit] Damping
Damping is the process by which sonic vibrations are converted into heat over time and distance. This can be achieved in several ways. For example, use of a material such as lead that is both heavy and soft, with the softness allowing it to damp the noise rather than allowing transmission. Making a sound wave transfer through different layers of material with different densities also assists in noise damping. This is the reason why open-celled foam is a good sound damper; the sound waves are forced to travel through multiple foam cells and their cell walls as sound travels through the foam medium. Improperly done, however, structural compliance can make things worse, enabling resonance. This process is analogous to a string holding wind-chimes: the string helps the chimes ring by isolating the vibration instead of damping it. Foam tapes may therefore be undependable in a soundproofing protocol.


[edit] Room Within A Room
A Room Within A Room (RWAR) is one method of isolating sound and stopping it transmitting to the outside world where it may be undesirable.

Most vibration / sound transfer from a room to the outside occurs through mechanical means. The vibration passes directly through the brick, woodwork and other solid structural elements. When it meets with an efficient sound board such as a wall, ceiling, floor or window, the vibration is amplified and heard in the second space. A mechanical transmission is much faster, more efficient and may be more readily amplified than an airborne transmission of the same initial strength.

The use of acoustic foams and other absorbent means are useless against this transmitted vibration. The user is required to break the connection between the room that contains the noise source and the outside world. This is called acoustic de-coupling. Ideal de-coupling involves eliminating vibration transfer in both solid materials and in the air, so air-flow into the room is often controlled. This has safety implications, for example proper ventilation must be assured and gas heaters cannot be used inside de-coupled space.

There are very successful professional products and methods available but such a construction is definitely within the reach of competent do-it-yourselfers.[2] Costs vary depending on the individual space, but it is clear that by doing it oneself, an individual can approximate the same result as professionals and save a substantial amount of money


[edit] Noise cancellation
Noise cancellation generators for active noise control are a relatively modern innovation. A microphone is used to pick up the sound that is then analyzed by a computer; then, sound waves with opposite polarity (not phase) are output through a speaker, causing destructive interference and cancelling much of the noise.


[edit] Noise barriers as exterior soundproofing
Main article: Noise barrier
Since the early 1970s it has become common practice in the United States (followed later by many other industrialized countries) to engineer noise barriers along major highways to protect adjacent residents from intruding roadway noise. The technology exists to predict accurately the optimum geometry for the noise barrier design. Noise barriers may be constructed of masonry, earth or a combination thereof. One of the earliest noise barrier designs was in Arlington, Virginia adjacent to Interstate 66, stemming from interests expressed by the Arlington Coalition on Transportation. Possibly the earliest scientifically designed and published noise barrier construction was in Los Altos, California in 1970.

Studio Monitor

The goal of most studio monitors is to produce a flat frequency response and a truthful representation of the source material. Unlike consumer speakers, which are often designed to make all audio material sound pleasing to the ear, the studio monitor generally attempts to paint an accurate audio image of the material with no unnatural emphasis or de-emphasis of particular frequencies. This is what it means when a monitor is said to be "flat". and "uncolored" or "transparent". Sound engineers usually require accurate sound reproduction from their speakers especially during audio mixing and mastering. This enables the engineer to mix a track that will sound consistent on high-end audio, low quality radios, in a club, in a car stereo or a home stereo. Some engineers, however, prefer to work with monitors that are far from accurate because they reflect the type of systems that end-users will be listening through.

Studio monitors can be active (they include one or more internal power amplifier(s)), or passive (they need an external power amplifier). Active models are usually bi-amplified.

Contents [hide]
1 History
2 Application
3 Home Audio versus Pro Audio
4 See also
5 References



[edit] History
In the early days of the recording industry, studio monitors were used primarily to check for noise interference and obvious technical problems rather than for making artistic evaluations of the performance and recording. Musicians were recorded live and the producer judged the performance on this basis, relying on simple tried-and-true microphone techniques to ensure that it had been adequately captured; playback through monitors was used simply to check that no obvious technical flaws had spoiled the original recording.

As a result, early monitors tended to be crude. The state of the art loudspeakers of the era were massive horn-loaded systems and were consequently used almost exclusively in cinemas. High-end loudspeaker design grew out of the demands of the motion picture industry and most of the early loudspeaker pioneers worked in Los Angeles where they attempted to solve the problems of cinema sound. Designing monitors for recording studios wasn’t a major priority.

The first high-quality loudspeaker developed expressly as a studio monitor was the Altec 604 Duplex in 1944. This innovative driver has historically been regarded as the work of James B. Lansing who’d previously supplied the drivers for the Shearer Horn in 1936, a speaker that had rapidly become the industry standard in motion-picture sound. He’d also designed the smaller Iconic and this was widely employed at the time as a motion-picture studio monitor. The 604 was a relatively compact coaxial design and within a few years it became the industry standard in the United States, a position it maintained in its various incarnations (the 604 went through eleven model-changes) over the next 25 years. It was common in US studios throughout the 1950s and 60s and remained in continuous production until 1998. In the UK, Tannoy introduced its own coaxial design, the Dual Concentric, and this assumed the same reference role in Europe as the Altec 604 held in the USA.

Monitor usage in the industry was highly conservative, with almost monopolistic reliance on industry “standards”, in spite of the sonic failings of these aging designs. The Altec 604 had a notoriously ragged frequency response but almost all U.S studios continued to use it because virtually every producer and engineer knew its sound intimately and were practiced at listening through its sonic limitations. Recording through unfamiliar monitors, no matter how technically advanced, was hazardous because engineers not used to their sonic signature could make poor production decisions and it was financially unviable to give production staff expensive studio time to familiarize themselves with new monitors. As a result, pretty well every U.S studio had a set of 604’s and every European studio a Tannoy Dual Concentric or two.

However, in 1959, at the height of its industry dominance, Altec made the mistake of replacing the 604 with the 605A Duplex, a design widely regarded as inferior to its predecessor. There was a backlash from some record companies and studios and this allowed Altec’s competitor, JBL (a company originally started by 604 designer James B. Lansing), to make inroads into the pro monitor market. Capitol Records quickly replaced their Altecs with JBL D50 Monitors and a few years later their UK affiliate, EMI, also made the move to JBL’s. Although Altec re-introduced the 604 as the "E" version Super Duplex in response to the criticism, they now had a major industry rival to contend with. Over the next decade most of the developments in studio monitor design originated from JBL.

As recording became less and less “live” and multi-tracking and overdubbing became the norm, the studio monitor became far more crucial to the recording process. When there was no original performance outside what existed on the tape, the monitor became the touchstone of all engineering and production decisions. As a result, accuracy and transparency became paramount and the conservatism evident in the retention of the 604 as the standard for over twenty years began to give way to fresh technological development. Despite this, the 604 continued to be widely used - mainly because many engineers and producers were so familiar with their sonic signature that they were reluctant to change. It wasn’t until 1975 that JBL overtook Altec as the monitor of choice for most studios.

In the late 1960’s JBL introduced two monitors which helped secure them preeminence in the industry. The 4320 was a direct competitor to the Altec 604 but was a more accurate and powerful speaker and it quickly made inroads against the industry standard. However, it was the more compact 4310 that revolutionized monitoring by introducing the idea of close or “nearfield” monitoring. (The sound field very close to a sound source is called the "near-field". By "very close" is meant in the predominantly direct, rather than reflected, soundfield. A near-field speaker is a compact studio monitor designed for listening at close distances (3’-5’), so, in theory, the effects of poor room acoustics are greatly reduced.)

The 4310 was small enough to be placed on the recording console and listened to from much closer distances than the traditional large wall-(or “soffit”) mounted main monitors. As a result, studio-acoustic problems were minimized. Smaller studios found the 4310 ideal and that monitor and its successor, the 4311, became studio fixtures throughout the 1970’s. Ironically, the 4310 had been designed to replicate the sonic idiosyncracies of the Altec 604 but in a smaller package to cater for the technical needs of the time. The 4311 was so popular with professionals that JBL introduced a domestic version for the burgeoning home-audio market. This speaker, the JBL L-100, (or "Century") was a massive success and became the biggest-selling hi-fi speaker ever within a few years.

The major studios continued to use huge designs mounted on the wall which were able to produce prodigious SPL’s and amounts of bass. This trend reached its zenith with The Who’s employment in their studio of a dozen JBL 4350 monitors, each capable of 125 dB and containing two fifteen-inch woofers and a twelve-inch mid-bass driver. Most studios, however, also used more modest monitoring devices to check how recordings would sound through car speakers and cheap home systems. A favourite “grot-box” monitor employed in this way was the Auratone 5C, a crude single-driver device that gave a reasonable facsimile of typical lo-fi sound.

However, a backlash against the Behemoth Monitor was soon to take place. With the advent of Punk, New Wave, Indie, and Lo-Fi, a reaction to high-tech recording and large corporate-style studios set in and do-it-yourself recording methods became the vogue. Smaller, less expensive, recording studios needed smaller, less expensive monitors and the Yamaha NS-10, a design introduced in 1978 ironically for the home audio market, became the monitor of choice for many studios in the 1980’s. A variety of stories, probably apocryphal, abound about why the NS-10 assumed this role but it gradually became an industry adage that “if it sounds good on the NS-10 it’ll sound good on anything”. While its sound-quality has often been derided, even by those who monitor through it, the NS-10 continues in use to this day and many more successful recordings have been produced with its aid over the past twenty five years than with any other monitor.

By the mid-80’s the near-field monitor had become pre-eminent. The larger studios still had large main-monitors mounted in (or on) the wall but they were now mere supplements to the small monitors sitting on the meter-bridge and were viewed as prestige items mainly there to “impress the clients” and occasionally check for low-bass anomalies. Favourite large monitors of the time were the Westlake and Urei 813, designs based on a highly modified version of the almost ageless Altec 604. Fostex "Laboratory Series" monitors were to be found in the finest studios but with increasing costs of manufacture, they became rare. The once dominant JBL fell into disfavour as its designs were identified with 70’s excess. The new studio landscape was bare and stripped-down and the large three or four-way monitors were hardly in keeping with this philosophy.

Yamaha eventually discontinued the NS-10 due to the lack of availability of the wood pulp used in the woofer. Even so, old NS-10’s still dot the studio landscape and at present it seems to be the last of the old style Industry Standards. No single monitor has emerged to become the fixtures that the Altec 604, the JBL 4311, and the Yamaha NS-10 were in their day. It now seems that every producer and engineer has their personal favourite monitor and developments in recording and monitor design have enabled this trend to continue. Personal recording studios have accelerated the move towards customization and individuality as the need for common industry standards is lessened. Monitors have become more and more compact and portable so it’s now feasible for producers to take their personal monitor choices with them to different recording venues.

The main post-NS10 trend has been the almost universal acceptance of powered monitors where the speaker enclosure contains the driving amplifiers. The old style passive monitors required outboard power amplifiers to drive them as well as speaker wire to connect them. Powered monitors, by contrast, are much more convenient and streamlined single units which in addition have a number of technical advantages. The interface between speaker and amplifier is optimized, with greater control and precision, and advances in amplifier design have reduced the size and weight of the electronics significantly. The result has been that passive monitors are now only a sidelight to the powered market and are in danger of being phased out completely.


[edit] Application
The studio monitor may be the single most important piece of equipment in the recording studio. This stems from the fact that every aspect of the recording process can only be heard - and evaluated – by listening through a monitor. Without it, recording is just guesswork.

As a result, production staff need to trust their monitors. The less confidence they have that a monitor is telling them what they need to know to make a correct decision the more random and time-consuming the recording process becomes. If a monitor fails to distinguish between two very different microphones or conceals distortion in the recording chain, poor production decisions are likely to be made. Producers, engineers, and mixers consequently insist on monitors that are either highly accurate and transparent or ones that they’re intimately familiar with. In the latter case, they learn to listen through the flaws, tend to be conservative, and often stick to a specific monitor even when it becomes technically outdated because changing to a new one requires a period of adjustment.

There are a variety of approaches to monitor choice and much “subjective” disagreement hinges on which of these methods production staff adopt. Some rigorously pursue accuracy and transparency, on the grounds that only an accurate monitor can tell the unvarnished truth and enable them to make completely trustworthy decisions. Others may insist that the systems used by the consumers are themselves far from accurate and that it’s more important that monitors reflect what these lo-fi stereos are doing to the recordings rather than pretend that it's being reproduced with perfect accuracy. Others believe that monitors should exaggerate recording flaws and make them work harder to make them sound good. Many more stick to monitors they know well, even though they’re fully aware of their limitations. Arguments about the “right” monitor to use still rage on, although there’s a growing acceptance that “what works for you” is the best adage to follow when choosing monitors. Producers and engineers have differing goals; some are producing high-quality recordings for expensive home systems while others are catering mainly for boomboxes and car radios. Their choices of monitor invariably take account of these facts.

Monitor use is thus in most cases geared to preparing the recording for end use: for making it sound good to the consumer. As a result, while accuracy has traditionally been considered to be the sine qua non of the studio monitor, in practice producers, engineers, and mixers are less concerned with accuracy per se than with how well the monitor translates, i.e., how well recordings made with its aid sound through a variety of playback systems, ranging from audiophile esoterica to boom-boxes. The link between accuracy and translatability is unclear, and the fact that the three most popular monitors of all time, the Altec 604, the JBL 4311, and the Yamaha NS-10, were all far from accurate may not be entirely coincidental.





[edit] Home Audio versus Pro Audio
While no rigid distinction exists between speakers intended for consumer use and those designed as studio monitors, there has been an increasing gulf between the two markets in practice. Whereas in the 1970’s the JBL 4311’s domestic equivalent, the L-100, was used in a large number of homes, and the Yamaha NS-10 also served both domestically and professionally during the 1980’s, there are no present-day equivalents. Professional companies such as Fostex, Genelec, Adam Audio, KRK Systems, Mackie, Klein and Hummel, Quested, PMC, and M & K sell almost exclusively to the professional monitor market while most of the consumer audio manufacturers confine themselves to supplying speakers for the home. Even companies that straddle both worlds, like Tannoy, Focal/JM Labs, Dynaudio, and JBL, tend to clearly separate their pro and consumer lines.

There are a number of reasons for this:

Domestic speakers are generally less rugged and unable to cope with the often extreme conditions encountered in the recording studio;
pro monitors are generally designed to be listened to from much shorter distances than home speakers;
pro monitors are generally powered while domestic speakers are almost always passive;
pro monitors are voiced to be less flattering to the source than domestic speakers are.
An illuminating indication of the difference between the two markets is the fact that the observation that “it makes everything sound great” is seen as a criticism in the studio monitor world! Monitors are selected because they ostensibly don’t flatter the material played through them and offer a “warts and all” presentation that makes it less likely for producers/engineers to approve unsatisfactory productions. Monitors are intended to err, if at all, on the side of harshness and aggressiveness rather than on papering over recording flaws whereas domestic speakers are often designed to make even mediocre material sound palatable.

For some reason, domestic speakers haven’t followed the professional move towards the active and powered. In audiophile circles, this is probably due to the fact that powered speakers tend to emphasize sonic qualities they find uncongenial, as well as a desire to select separate components rather than simplify the audio chain. For this reason, the seemingly inevitable move to domestic powered speakers is more likely to come at the lower end of the consumer market.

Stereo Imaging

Stereo imaging is the audio jargon term used for that aspect of sound recording and reproduction concerning spatial locations of the performers, both laterally and in depth. An image is 'good' if the performers can be effortlessly located; 'bad' if there is no hope of doing so. A well-made stereo recording, properly reproduced, can provide good imaging within the front quadrant; a well-made Ambisonic recording, properly reproduced, can offer good imaging all around the listener and even including height information.

For many listeners, good imaging adds markedly to the pleasure of reproduced music. One may speculate that this is due to the evolutionary importance to humans of knowing where sounds are coming from, and that imaging may therefore be more important than some purely esthetic considerations in satisfying the listener. Listeners do exist who have difficulty paying attention to the musical content of a recording if the imaging is not good.

The quality of the imaging arriving at the listener's ear depends on numerous factors, of which the most important is the original "miking", that is, the choice and arrangement of the recording microphones (where "choice" refers here not to the brands chosen, but to the size and shape of the microphone diaphragms, and "arrangement" refers to microphone placement and orientation relative to other microphones). This is partly because miking simply affects imaging more than any other factor, and because, if the miking spoils the imaging, nothing later in the chain can recover it.

If miking is done well, then quality of imaging can be used to evaluate components in the record/playback chain (remembering that once the imaging is destroyed, it cannot be recovered).

It is worth noting that only a handful of recordings are miked for optimal imaging, and what usually passes for stereo, while being two-channel recording, is not true stereo because the imaging information is incoherent.

Imaging is usually thought of in the context of recording with two or more channels, though single-channel recording may convey depth information convincingly, and at least one expert thinks it can convey information about lateral placement, also.

SAE Institute

SAE Institute
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The SAE Institute (SAE in short, formerly also known as the School of Audio Engineering and the SAE Technology College) is a private college founded in 1976 by Sound Engineer/Record Producer Tom Misner. The first school was opened 1977 in Sydney, (Australia). SAE Institute offers courses in Audio engineering, 3D Animation, Multimedia and Digital Filmmaking. It is the largest college worldwide in these fields, and currently has campuses / facilities in 46 cities in 21 countries. Since 1998 SAE is additionally offering full university degrees through its global partnership with Middlesex University, London, United Kingdom, and in Australia already since 1996 through its co-operation with Southern Cross University.



[edit] History
1976 The School of Audio Engineering (SAE) is set up in June 1976 by Tom Misner[1] who, in so doing, develops the first practical/theoretical curriculum.
1977 The first 9 month course commences in February, in Byron Bay Australia with a Sony 4-track tape recorder and a custom made 12 channel mixing console.
1978 SAE Melbourne is established with a small 8 track studio and some editing tape machines.
1980 SAE Brisbane is established and SAE's first commercial studio Central Recorder is opened in Sydney.
1981 SAE Sydney commences with the acceptance of overseas students and gains the first form of government recognition - a public service grading.
1982 SAE Adelaide commences operation early in the year. SAE Perth is established in August.
1984 SAE Coffs Harbour (Australia) is set up for one year only to conduct a course on behalf of the local television and radio stations, but the college stays open for a further year.
1985 Resulting from a business trip to London, Tom Misner establishs SAE London, the first overseas college. Operation commences in March.
1986 SAE Munich, the first foreign language SAE school, opens in Germany. SAE Frankfurt commences late in the year.
1987 SAE Vienna the first college in Austria is opened in February. The first custom-designed program is conducted by SAE Germany for radio station 'Radio Free Europe'.
1988 SAE Berlin the third German college is opened late in the year with special assistance from the German government. A new course 'Live Sound Engineering' is offered by SAE.
1989 SAE introduces the 6 months part-time 'Studio Sound Certificate' course.
1990 SAE Auckland is established and is granted full government approval (New Zealand Qualifications Authority). SAE introduces the 'DJ and Sampling' course in London. SAE Glasgow commences operation late in the year.
1991 SAE Amsterdam, the new European operational head office is opened. One of the largest orders of Neve audio editing consoles ever placed (11 VR series consoles). SAE Singapore commences in September as the first audio college in Asia.
1992 The audio engineering programme is extended to 15 months part-time in all SAE colleges. SAE Kuala Lumpur opens in October.
1993 SAE Paris opens. Partial government approval and funding for students is given to SAE by the Government of France. SAE Hamburg opens and commences operation as the first SAE college to teach extensive digital practice. The studios are based upon the Soundtracs Jade console, Sony APR multitrack and ProTools III.
1994 SAE Kuala Lumpur gains government approval (Ministry of Education). The first SAE Book (Practical Studio Techniques by Tom Misner) is published. SAE forms an official link with the Australian Southern Cross University to offer a joint degree program (BA Music Production). The School of Audio Engineering changes its name to SAE Technology College. Tom Misner opens the only new large commercial recording studio in Australia to be built in the 90's, Mirage Studios.
1995 The first SAE-ProSchool is established in London teaching the Digidesign ProTools system. SAE Stockholm (Sweden) commences. SAE Zürich has now been established in the Technopark industrial complex. SAE Hobart, the 6th college in Australia, opens. SAE Cologne, the fifth German College, opens later in that year and offers both the audio and multimedia programmes. SAE Singapore receives government approval and is able to accept overseas students.
1996 The multimedia program is expanded to Zürich and Singapore. The first full university degree programme is launched by SAE Sydney with the co-operation of the Southern Cross University. SAE Frankfurt now offers the first live sound program in Germany. SAE Milano opens in Italy.
1997 Expansion of several SAE campuses. SAE forms the SAE Entertainment Company for professional production of CD ROM, CD extra, CD audio and internet homepages.
1998 SAE New York City is licensed. SAE Athens, Greece opens late in the year. SAE enters into a collaborative arrangement with Middlesex University, England and the first BA (Hons) degree programmes are run at the London, Munich and Sydney campuses.
1999 SAE Nashville starts operation. SAE purchases recording facility Studio 301 in Australia. SAE Munich starts the Digital Film Programme in November 1999, with Cologne, Hamburg and Vienna to follow in spring.
2000 SAE Munich starts Digital Film Arts Degree. SAE Hamburg starts the Digital Film Program. Four franchise schools are established in India. SAE Frankfurt, Amsterdam, Stuttgart, and Berlin areexpanded.
2001 sees the opening of SAE Miami, Liverpool and Madrid. Tom Misner purchases the largest recording studio in Germany, which is now part of the Studios 301 Group.
2002 SAE Adelaide and SAE Perth turn 20. The new Digital Film Making Program is starting in Australia and Europe. SAE Thiruvananthapuram, India, commences operation. SAE Berlin and SAE Athens are approved as Degree Centres by Middlesex University, England. SAE Bangkok starts operation.
2003 The SAE Alumni Association is founded. SAE Brussels, Belgium, SAE SAE Institute Bangkok and SAE Yangon, Myanmar, are opened. SAE Berlin is offering the first Bachelor courses. The new headquarters in Byron Bay, Australia opens.
2004 SAE Munich, Amsterdam, Melbourne, Stuttgart and Hamburg all move to new, bigger and improved premises. SAE Leipzig opens. SAE Barcelona opens. SAE acquires QANTM, Australia's leading production, new media and training company.
2005 Tom Misner takes over console maker AMS Neve. SAE Dubai, first college in the Middle East, opens. SAE Los Angeles, fourth college in the US, opens. First SAE Alumni worldwide conference is held in Frankfurt.
2006 SAE opens its second Middle East campus in Kuwait.

[edit] External links
SAE Institute for Audio Engineering, Filmmaking, Animation and Multimedia & Web Design Courses
SAE Institute Byron Bay - International Headquarters offering Audio Engineering, Filmmaking and Multimedia & Web Design Courses
SAE Institute UK
SAE Institute Germany
SAE Institute Austria
SAE Institute Italia
SAE Institute France
SAE Institute Spain
SAE Institute Bangkok
SAE Institute Malaysia
SAE Institute Kuwait
SAE Institute Singapore
SAE Institute The Netherlands

Microphone

A microphone, sometimes referred to as a mike or mic (both IPA pronunciation: [maɪk]), is an acoustic to electric transducer or sensor that converts sound into an electrical signal.


A Neumann U87 capacitor microphoneMicrophones are used in many applications such as telephones, tape recorders, hearing aids, motion picture production, live and recorded audio engineering, in radio and television broadcasting and in computers for recording voice, VoIP, and for non-acoustic purposes such as ultrasonic checking.

Contents [hide]
1 History
2 Principle of operation
3 Microphone varieties
3.1 Condenser, capacitor or electrostatic microphones
3.1.1 Technology
3.1.1.1 DC-biased microphone operating principle
3.1.1.2 RF condenser microphone operating principle
3.1.2 Usage
3.1.3 Electret condenser microphones
3.2 Dynamic microphones
3.2.1 Moving coil microphones
3.2.1.1 Technology
3.2.2 Ribbon microphones
3.3 Carbon microphones
3.4 Piezo microphones
3.4.1 Technology
3.4.2 Usage
3.5 Laser microphones
3.5.1 Usage
3.6 Liquid microphones
3.6.1 Technology
3.6.2 Usage
3.7 MEMS microphones
3.8 Speakers as microphones
4 Capsule design and directivity
5 Microphone polar patterns
6 Application-specific microphone designs
7 Connectivity
7.1 Connectors
7.2 Impedance matching
8 Measurements and specifications
9 Measurement microphones
9.1 Microphone calibration techniques
9.1.1 Pistonphone apparatus
9.1.2 Reciprocal method
10 Microphone array and array microphones
11 See also
12 Microphone manufacturers
13 External links
14 References



[edit] History
Several early inventors built primitive microphones (then called transmitters) prior to Alexander Bell, but the first commercially practical microphone was the carbon microphone conceived in October 1876 by Thomas Edison. Many early developments in microphone design took place at Bell Laboratories. See also Timeline of the telephone.


[edit] Principle of operation

An Oktava condenser microphone.A microphone is a device made to capture waves in air, water (hydrophone) or hard material and translate them into an electrical signal. The most common method is via a thin membrane producing some proportional electrical signal. Most microphones in use today for audio use electromagnetic generation (dynamic microphones), capacitance change (condenser microphones) or piezoelectric generation to produce the signal from mechanical vibration. The piezoelectric microphone is now largely obsolete. However, piezoelectric pickups are still the most common device for amplifying acoustic guitars, usually placed under the guitar's saddle or embedded in the bridge.


[edit] Microphone varieties

[edit] Condenser, capacitor or electrostatic microphones

Inside the Oktava 319 condenser microphone.
[edit] Technology
In a condenser microphone, also known as a capacitor microphone, the diaphragm acts as one plate of a capacitor, and the vibrations produce changes in the distance between the plates.

There are two methods of extracting an audio output from the transducer thus formed. They are known as DC biased and RF (or HF) condenser microphones.


[edit] DC-biased microphone operating principle
The plates are biased with a fixed charge (Q). The voltage maintained across the capacitor plates changes with the vibrations in the air, according to the capacitance equation:


where Q = charge in coulombs, C = capacitance in farads and V = potential difference in volts. The capacitance of the plates is inversely proportional to the distance between them for a parallel-plate capacitor. (See capacitance for details.)

A nearly constant charge is maintained on the capacitor. As the capacitance changes, the charge across the capacitor does change very slightly, but at audible frequencies it is sensibly constant. The capacitance of the capsule and the value of the bias resistor form a filter which is highpass for the audio signal, and lowpass for the bias voltage. Note that the time constant of a RC circuit equals the product of the resistance and capacitance.

Within the time-frame of the capacitance change (on the order of 100 μs), the charge thus appears practically constant and the voltage across the capacitor adjusts itself instantaneously to reflect the change in capacitance. The voltage across the capacitor varies above and below the bias voltage. The voltage difference between the bias and the capacitor is seen across the series resistor. The voltage across the resistor is amplified for performance or recording.


[edit] RF condenser microphone operating principle
In a DC-biased condenser microphone, a high capsule polarisation voltage is necessary. In contrast, RF condenser microphones use a comparatively low RF voltage, generated by a low-noise oscillator. The oscillator is frequency modulated by the capacitance changes produced by the sound waves moving the capsule diaphragm. Demodulation yields a low-noise audio frequency signal with a very low source impedance. This technique achieves better low frequency response - in fact it will theoretically operate down to DC.

The RF biasing process results in a lower electrical impedance capsule, a useful byproduct of which is that RF condenser microphones can be operated in damp weather conditions which would effectively short out a DC biased microphone. The Sennheiser "MKH" series of microphones use the RF biased technique.


[edit] Usage
Condenser microphones span the range from cheap throw-aways to high-fidelity quality instruments. They generally produce a high-quality audio signal and are now the popular choice in laboratory and studio recording applications. They require a power source, provided either from microphone inputs as phantom power or from a small battery. Professional microphones often sport an external power supply for reasons of quality perception. Power is necessary for establishing the capacitor plate voltage, and is also needed for internal amplification of the signal to a useful output level. Condenser microphones are also available with two diaphragms, the signals from which can be electrically connected such as to provide a range of polar patterns (see below), such as cardioid, omnidirectional and figure-eight. It is also possible to vary the pattern smoothly with some microphones, for example the Røde NT2000.


[edit] Electret condenser microphones
Main article: Electret microphone
An electret microphone is a relatively new type of capacitor microphone invented at Bell laboratories in 1962 by Gerhard Sessler and Jim West[1]. An electret is a dielectric material that has been permanently electrically charged or polarized. The name comes from electrostatic and magnet; a static charge is embedded in an electret by alignment of the static charges in the material, much the way a magnet is made by aligning the magnetic domains in a piece of iron. They are used in many applications, from high-quality recording and lavalier use to built-in microphones in small sound recording devices and telephones. Though electret microphones were once low-cost and considered low quality, the best ones can now rival capacitor microphones in every respect and can even offer the long-term stability and ultra-flat response needed for a measuring microphone. Unlike other capacitor microphones, they require no polarizing voltage, but normally contain an integrated preamplifier which does require power (often incorrectly called polarizing power or bias). This preamp is frequently phantom powered in sound reinforcement and studio applications. While few electret microphones rival the best DC-polarized units in terms of noise level, this is not due to any inherent limitation of the electret. Rather, mass production techniques needed to produce electrets cheaply don't lend themselves to the precision needed to produce the highest quality microphones.


[edit] Dynamic microphones
Dynamic microphones work via electromagnetic induction. They are robust, relatively inexpensive and resistant to moisture, and for this reason they are widely used on-stage by singers. Dynamic microphones are velocity receivers. There are two basic types: the moving coil microphone and the ribbon microphone.


[edit] Moving coil microphones

The Shure SM57 and Beta 57A dynamic microphones
[edit] Technology
A small movable induction coil, positioned in the magnetic field of a permanent magnet, is attached to the diaphragm. When sound enters through the windscreen of the microphone, the sound wave moves the diaphragm. When the diaphragm vibrates, the coil moves in the magnetic field, producing a varying current in the coil through electromagnetic induction. A single dynamic membrane will not respond linearly to all audio frequencies. Some microphones for this reason utilize multiple membranes for the different parts of the audio spectrum and then combine the resulting signals. Combining the multiple signals correctly is difficult and designs that do this are rare and tend to be expensive. There are on the other hand several designs that are more specifically aimed towards isolated parts of the audio spectrum. AKG D112 is for example designed for bass content rather than treble. In audio engineering several kinds of microphones are often used at the same time to get the best result.

The dynamic principle is exactly the same as in a loudspeaker, only reversed.


[edit] Ribbon microphones
Main article: Ribbon microphone
In ribbon microphones a thin, usually corrugated metal ribbon is suspended in a magnetic field. The ribbon is electrically connected to the microphone's output, and its vibration within the magnetic field generates the electrical signal. Ribbon microphones are similar to moving coil microphones in the sense that both produce sound by means of magnetic induction. Basic ribbon microphones detect sound in a bidirectional (also called figure-eight) pattern because the ribbon, which is open to sound both front and back, responds to the pressure gradient rather than the sound pressure. Though the symmetrical front and rear pickup can be a nuisance in normal stereo recording, the high side rejection can be used to advantage by positioning a ribbon microphone horizontally, for example above cymbals, so that the rear lobe picks up only sound from the cymbals. Other directional patterns are produced by enclosing one side of the ribbon in an acoustic trap or baffle, allowing sound to reach only one side. Ribbon microphones give very high quality sound reproduction, and were once valued for this reason, but a good low-frequency response can be obtained only if the ribbon is suspended very loosely, and this makes them fragile. Protective wind screens can reduce the danger of damaging the ribbon, but will somewhat reduce the treble response.

Ribbon microphones don't require phantom power; in fact, this voltage can damage these microphones.


[edit] Carbon microphones
Main article: Carbon microphone
A carbon microphone, formerly used in telephone handsets, is a capsule containing carbon granules pressed between two metal plates. A voltage is applied across the metal plates, causing a small current to flow through the carbon. One of the plates, the diaphragm, vibrates in sympathy with incident sound waves, applying a varying pressure to the carbon. The changing pressure deforms the granules, causing the contact area between each pair of adjacent granules to change, and this causes the electrical resistance of the mass of granules to change. The changes in resistance cause a corresponding change in the voltage across the two plates, and hence in the current flowing through the microphone, producing the electrical signal. Carbon microphones were once commonly used in telephones; they have extremely low-quality sound reproduction and a very limited frequency response range, but are very robust devices.

Unlike other microphone types, the carbon microphone can also be used as a type of amplifier, using a small amount of sound energy to produce a larger amount of electrical energy. Carbon microphones found use as early telephone repeaters, making long distance phone calls possible in the era before vacuum tubes. These repeaters worked by mechanically coupling a magnetic telephone receiver to a carbon microphone: the faint signal from the receiver was transferred to the microphone, with a resulting stronger electrical signal to send down the line. (One illustration of this amplifier effect was the oscillation caused by feedback, resulting in an audible squeal from the old "candlestick" telephone if its earphone was placed near the carbon microphone.)


[edit] Piezo microphones

[edit] Technology
A piezo microphone uses the phenomenon of piezoelectricity—the ability of some materials to produce a voltage when subjected to pressure—to convert vibrations into an electrical signal. An example of this is Rochelle salt (potassium sodium tartrate), which is a piezoelectric crystal that works as a transducer, both as a microphone and as a slimline loudspeaker component.


[edit] Usage
Piezo transducers are often used as contact microphones to amplify sound from acoustic musical instruments, or to record sounds in unusual environments (underwater, for instance). Saddle mounted pickups on acoustic guitars are generally piezos that are mechanically connected to the strings through the saddle. This type of microphone is not to be confused with magnetic coil pickups commonly visible on typical electric guitars.


[edit] Laser microphones

[edit] Usage
Laser microphones are new, very rare and expensive, and are most commonly portrayed in movies as spying devices.


[edit] Liquid microphones

[edit] Technology
Early microphones did not produce intelligible speech, until Alexander Graham Bell made a set of improvements. Bell’s liquid transmitter consisted of a metal cup filled with dilute sulfuric acid. A sound wave caused the diaphragm to move, forcing a brass tube to move up and down in the liquid. The electrical resistance between the wire and the cup was then inversely proportional to the length of wire submerged. Elisha Gray filed a patent for a version using a needle instead of the brass tube. Other minor variations and improvements were made to the liquid microphone by Majoranna, Chambers, Vanni, Sykes, and Elisha Gray, and one version was even patented by Reginald Fessenden in 1903.


[edit] Usage
These were the first working microphones, but they were not practical for commercial application and are utterly obsolete now. It was with a liquid transmitter that the famous first phone conversation between Bell and Watson took place. Other inventors soon devised superior devices.


[edit] MEMS microphones
The MEMS microphone is also called a microphone chip or silicon microphone. The pressure-sensitive diaphragm is etched directly on a silicon chip by MEMS techniques[citation needed], and is usually accompanied with integrated preamplifier. Most MEMS microphones are modern embodiments of the standard condenser microphone. Often MEMS mics have a built in ADC on the same CMOS chip making the chip a digital microphone and easily integrated into modern digital products. Major manufacturers using MEMS manufacturing for silicon microphones are Akustica (AKU200x), Infineon (SMM310 product), Knowles Electronics and Sonion MEMS.


[edit] Speakers as microphones
A loudspeaker, a transducer that turns an electrical signal into sound waves, is the functional opposite of a microphone. Since a conventional speaker is constructed much like a dynamic microphone (with a diaphragm, coil and magnet), speakers can actually work "in reverse" as microphones. The result, though, is a microphone with poor quality, limited frequency response (particularly at the high end), and poor sensitivity.

In practical use, speakers are sometimes used as microphones in such applications as intercoms or walkie-talkies, where high quality and sensitivity are not needed. However, there is at least one other novel application of this principle; using a medium-size woofer placed closely in front of a "kick" (bass drum) in a drum set to act as a microphone. This has been commercialized with the Yamaha "Subkick".[1]


[edit] Capsule design and directivity
The shape of the microphone defines its directivity. Inner elements are of major importance and concerns the structural shape of the capsule, outer elements may be the interference tube.

A pressure gradient microphone is a microphone in which both sides of the diaphragm are exposed to the incident sound and the microphone is therefore responsive to the pressure differential (gradient) between the two sides of the membrane. Sound incident parallel to the plane of the diaphragm produces no pressure differential, giving pressure-gradient microphones their characteristic figure-eight directional patterns.

The capsule of a pressure microphone however is closed on one side, which results in an omnidirectional pattern.


[edit] Microphone polar patterns
Common polar patterns for microphones (Microphone facing top of page in diagram, parallel to page):

Omnidirectional



Subcardioid



Cardioid



Supercardioid




Hypercardioid



Bi-directional



Shotgun





A microphone's directionality or polar pattern indicates how sensitive it is to sounds arriving at different angles about its central axis. The above polar patterns represent the locus of points that produce the same signal level output in the microphone if a given sound pressure level is generated from that point. How the physical body of the microphone is oriented relative to the diagrams depends on the microphone design. For large-membrane microphones such as in the Oktava (pictured above), the upward direction in the polar diagram is usually perpendicular to the microphone body, commonly known as "side fire". For small diaphragm microphones such as the Shure (also pictured above), it usually extends from the axis of the microphone commonly known as "end fire". Some microphone designs combine several principles in creating the desired polar pattern. This ranges from shielding (meaning diffraction/dissipation/absorption) by the housing itself to electronically combining dual membranes.

An omnidirectional microphone's response is generally considered to be a perfect sphere in three dimensions. In the real world, this is not the case. As with directional microphones, the polar pattern for an "omnidirectional" microphone is a function of frequency. The body of the microphone is not infinitely small and, as a consequence, it tends to get in its own way with respect to sounds arriving from the rear, causing a slight flattening of the polar response. This flattening increases as the diameter of the microphone (assuming it's cylindrical) reaches the wavelength of the frequency in question. Therefore, the smallest diameter microphone will give the best omnidirectional characteristics at high frequencies. The wavelength of sound at 10 kHz is little over an inch (3.4 cm) so the smallest measuring microphones are often 1/4" (6 mm) in diameter, which practically eliminates directionality even up to the highest frequencies. Omnidirectional microphones, unlike cardioids, do not employ resonant cavities as delays, and so can be considered the "purest" microphones in terms of low coloration; they add very little to the original sound. Being pressure-sensitive they can also have a very flat low-frequency response down to 20 Hz or below. Pressure-sensitive microphones also respond much less to wind noise than directional (velocity sensitive) microphones.

A unidirectional microphone is sensitive to sounds from only one direction. The diagram above illustrates a number of these patterns. The microphone faces upwards in each diagram. The sound intensity for a particular frequency is plotted for angles radially from 0 to 360°. (Professional diagrams show these scales and include multiple plots at different frequencies. These diagrams just provide an overview of the typical shapes and their names.)

The most common unidirectional microphone is a cardioid microphone, so named because the sensitivity pattern is heart-shaped (see cardioid). A hyper-cardioid is similar but with a tighter area of front sensitivity and a tiny lobe of rear sensitivity. These two patterns are commonly used as vocal or speech microphones, since they are good at rejecting sounds from other directions. Because they employ internal cavities to provide front-back delay, directional microphones tend to have more coloration than omnis, and they also suffer from low-frequency roll-off. These problems are overcome to a large extent by careful design, but only the best cardioids can begin to approach the performance of a tiny low-cost omni in terms of absolute accuracy. This is not always recognised, but is the price paid for directionality, often needed to exclude ambient reverberation wherever very close placement is impossible.

Figure 8 or bi-directional microphones receive sound from both the front and back of the element. Most ribbon microphones are of this pattern.


An Audio-Technica shotgun microphoneShotgun microphones are the most highly directional. They have small lobes of sensitivity to the left, right, and rear but are significantly more sensitive to the front. This results from placing the element inside a tube with slots cut along the side; wave-cancellation eliminates most of the off-axis noise. Shotgun microphones are commonly used on TV and film sets, and for field recording of wildlife.

An omnidirectional microphone is a pressure transducer; the output voltage is proportional to the air pressure at a given time.

On the other hand, a figure-8 pattern is a pressure gradient transducer; the output voltage is proportional to the difference in pressure on the front and on the back side. A sound wave arriving from the back will lead to a signal with a polarity opposite to that of an identical sound wave from the front. Moreover, shorter wavelengths (higher frequencies) are picked up more effectively than lower frequencies.

A cardioid microphone is effectively a superposition of an omnidirectional and a figure-8 microphone; for sound waves coming from the back, the negative signal from the figure-8 cancels the positive signal from the omnidirectional element, whereas for sound waves coming from the front, the two add to each other. A hypercardioid microphone is similar, but with a slightly larger figure-8 contribution.

Since pressure gradient transducer microphones are to some extent directional, their frequency response is dependent on the distance to the sound source. This is known as the proximity effect, resulting in a bass boost at distances of a few centimeters[citation needed].


[edit] Application-specific microphone designs
A lavalier microphone is made for hands-free operation. These small microphones are worn on the body and held in place either with a lanyard worn around the neck or a clip fastened to clothing. The cord may be hidden by clothes and either run to an RF transmitter in a pocket or clipped to a belt (for mobile use), or run directly to the mixer (for stationary applications).

A wireless microphone is one which does not use a cable. It usually transmits its signal using a small FM radio transmitter to a nearby receiver connected to the sound system, but it can also use infrared light if the transmitter and receiver are within sight of each other.

A contact microphone is designed to pick up vibrations directly from a solid surface or object, as opposed to sound vibrations carried through air. One use for this is to detect sounds of a very low level, such as those from small objects or insects. The microphone commonly consists of a magnetic (moving coil) transducer, contact plate and contact pin. The contact plate is placed against the object from which vibrations are to be picked up; the contact pin transfers these vibrations to the coil of the transducer. Contact microphones have been used to pick up the sound of a snail's heartbeat and the footsteps of ants. A portable version of this microphone has recently been developed.

A throat microphone is a variant of the contact microphone, used to pick up speech directly from the throat, around which it is strapped. This allows the device to be used in areas with ambient sounds that would otherwise make the speaker inaudible.

A parabolic microphone uses a parabolic reflector to collect and focus sound waves onto a microphone receiver, in much the same way that a parabolic antenna (e.g. satellite dish) does with radio waves. Typical uses of this microphone, which has unusually focused front sensitivity and can pick up sounds from many meters away, include nature recording, outdoor sporting events, eavesdropping, law enforcement, and even espionage. Parabolic microphones are not typically used for standard recording applications, because they tend to have poor low-frequency response as a side effect of their design.


[edit] Connectivity

[edit] Connectors
The most common connectors used by microphones are:

Male XLR connector on professional microphones
¼ inch mono phone plug on less expensive consumer microphones
3.5 mm (Commonly referred to as 1/8 inch mini) mono mini phone plug on very inexpensive and computer microphones
Some microphones use other connectors, such as 1/4 inch TRS (tip ring sleeve), 5-pin XLR, or stereo mini phone plug (1/8 inch TRS) on some stereo microphones. Some lavalier microphones use a proprietary connector for connection to a wireless transmitter. Since 2005, professional-quality microphones with USB connections have begun to appear, designed for direct recording into computer-based software studios.


[edit] Impedance matching
Microphones have an electrical characteristic called impedance, measured in ohms (Ω) that depends on the design. Typically, the rated impedance is stated.[2] Low impedance is considered under 600 Ω. Medium impedance is considered between 600 Ω and 10 kΩ. High impedance is above 10 kΩ. Most professional microphones are low impedance, about 200 Ω or lower. Low-impedance microphones are preferred over high impedance for two reasons: one is that using a high-impedance microphone with a long cable will result in loss of high frequency signal due to the capacitance of the cable; the other is that long high-impedance cables tend to pick up more hum (and possibly radio-frequency interference (RFI) as well). However, some equipment, such as vacuum tube guitar amplifiers, has an input impedance that is inherently high, requiring the use of a high impedance microphone or a matching transformer. Nothing will be damaged if the impedance between microphone and other equipment is mismatched; the worst that will happen is a reduction in signal or change in frequency response.

To get the best sound in most cases, the impedance of the microphone must be distinctly lower (by a factor of at least five) than that of the equipment to which it is connected. Most microphones are designed not to have their impedance "matched" by the load to which they are connected; doing so can alter their frequency response and cause distortion, especially at high sound pressure levels. There are transformers (confusingly called matching transformers) that adapt impedances for special cases such as connecting microphones to DI units or connecting low-impedance microphones to the high-impedance inputs of certain amplifiers, but microphone connections generally follow the principle of bridging (voltage transfer), not matching (power transfer). In general, any XLR microphone can usually be connected to any mixer with XLR microphone inputs, and any plug microphone can usually be connected to any jack that is marked as a microphone input, but not to a line input. This is because the signal level of a microphone is typically 40-60 dB lower (a factor of 100 to 1000) than a line input. Microphone inputs include the necessary amplification circuitry to deal with these very low level signals. The exception to these comments is in the case of certain ribbon and dynamic microphones which are most linear when operated into a load of known impedance [3]


[edit] Measurements and specifications

A comparison of the far field on-axis frequency response of the Oktava 319 and the Shure SM58Because of differences in their construction, microphones have their own characteristic responses to sound. This difference in response produces non-uniform phase and frequency responses. In addition, microphones are not uniformly sensitive to sound pressure, and can accept differing levels without distorting. Although for scientific applications microphones with a more uniform response are desirable, this is often not the case for music recording, as the non-uniform response of a microphone can produce a desirable coloration of the sound. There is an international standard for microphone specifications,[4] but few manufacturers adhere to it. As a result, comparison of published data from different manufacturers is difficult because different measurement techniques are used. The Microphone Data Website has collated the technical specifications complete with pictures, response curves and technical data from the microphone manufacturers for every currently listed microphone, and even a few obsolete models, and shows the data for them all in one common format for ease of comparison.[2]. Caution should be used in drawing any solid conclusions from this or any other published data, however, unless it is known that the manufacturer has supplied specifications in accordance with IEC 60268-4.

A frequency response diagram plots the microphone sensitivity in decibels over a range of frequencies (typically at least 0–20 kHz), generally for perfectly on-axis sound (sound arriving at 0° to the capsule). Frequency response may be less informatively stated textually like so: "30 Hz–16 kHz ±3 dB". This is interpreted as a (mostly) linear plot between the stated frequencies, with variations in amplitude of no more than plus or minus 3 dB. However, one cannot determine from this information how smooth the variations are, nor in what parts of the spectrum they occur. Note that commonly-made statements such as "20 Hz–20 kHz" are meaningless without a decibel measure of tolerance. Directional microphones' frequency response varies greatly with distance from the sound source, and with the geometry of the sound source. IEC 60268-4 specifies that frequency response should be measured in plane progressive wave conditions (very far away from the source) but this is seldom practical. Close talking microphones may be measured with different sound sources and distances, but there is no standard and therefore no way to compare data from different models unless the measurement technique is described.

The self-noise or equivalent noise level is the sound level that creates the same output voltage as the microphone does in the absence of sound. This represents the lowest point of the microphone's dynamic range, and is particularly important should you wish to record sounds that are quiet. The measure is often stated in dB(A), which is the equivalent loudness of the noise on a decibel scale frequency-weighted for how the ear hears, for example: "15 dBA SPL" (SPL means sound pressure level relative to 20 micropascals). The lower the number the better. Some microphone manufacturers state the noise level using ITU-R 468 noise weighting, which more accurately represents the way we hear noise, but gives a figure some 11 to 14 dB higher. A quiet microphone will measure typically 20 dBA SPL or 32 dB SPL 468-weighted.

The maximum SPL (sound pressure level) the microphone can accept is measured for particular values of total harmonic distortion (THD), typically 1%. This is generally inaudible, so one can safely use the microphone at this level without harming the recording. Example: "142 dB SPL peak (<1% THD)". The higher the value, the better, although microphones with a very high maximum SPL also have a higher self-noise.

The clipping level is perhaps a better indicator of maximum usable level, as the 1% THD figure usually quoted under max SPL is really a very mild level of distortion, quite inaudible especially on brief high peaks. Harmonic distortion from microphones is usually of low-order (mostly third harmonic) type, and hence not very audible even at 3-5%. Clipping, on the other hand, usually caused by the diaphragm reaching its absolute displacement limit (or by the preamplifier), will produce a very harsh sound on peaks, and should be avoided if at all possible. For some microphones the clipping level may be much higher than the max SPL.

The dynamic range of a microphone is the difference in SPL between the noise floor and the maximum SPL. If stated on its own, for example "120 dB", it conveys significantly less information than having the self-noise and maximum SPL figures individually.

Sensitivity indicates how well the microphone converts acoustic pressure to output voltage. A high sensitivity microphone creates more voltage and so will need less amplification at the mixer or recording device. This is a practical concern but is not directly an indication of the mic's quality, and in fact the term sensitivity is something of a misnomer, 'transduction gain' being perhaps more meaningful, (or just "output level") because true sensitivity will generally be set by the noise floor, and too much "sensitivity" in terms of output level will compromise the clipping level. There are two common measures. The (preferred) international standard is made in millivolts per pascal at 1 kHz. A higher value indicates greater sensitivity. The older American method is referred to a 1 V/Pa standard and measured in plain decibels, resulting in a negative value. Again, a higher value indicates greater sensitivity, so −60 dB is more sensitive than −70 dB.


[edit] Measurement microphones
Some microphones are intended for use as standard measuring microphones for the testing of speakers and checking noise levels etc. These are calibrated transducers and will usually be supplied with a calibration certificate stating absolute sensitivity against frequency.


[edit] Microphone calibration techniques

[edit] Pistonphone apparatus
A pistonphone is an acoustical calibrator (sound source) using a closed coupler to generate a precise sound pressure for the calibration of instrumentation microphones. The principle relies on a piston mechanically driven to move at a specified rate on a fixed volume of air to which the microphone under test is exposed. The air is assumed to be compressed adiabatically and the SPL in the chamber can be calculated from PV = const. The pistonphone method only works at low frequencies, but it can be accurate and yields an easily calculable sound pressure level. The standard test frequency is usually around 250 Hz.


[edit] Reciprocal method
This method relies on the reciprocity of one or more microphones in a group of 3 to be calibrated. It can still be used when only one of the microphones is reciprocal (exhibits equal response when used as a microphone or as a loudspeaker).

Loudspeaker

A loudspeaker, speaker, or speaker system is an electromechanical transducer which converts an electrical signal into sound. The term loudspeaker is currently used for both individual devices and for complete systems consisting of one or more drivers (as the individual transducers are often called) in an enclosure, often with a crossover circuit. Their cost may range from pennies in a cheap radio to high-fidelity speaker systems costing many thousands of dollars. Loudspeakers are the most variable elements in any audio system, regardless of cost, and are responsible for marked audible differences between otherwise identical sound systems.

Full-range speaker systems are typically multi-driver systems, particularly when high SPL output or high accuracy are required. "Multi driver" means a speaker system containing two or more drive units, possibly including woofers, midranges, tweeters, or supertweeters. In loudspeaker specifications, systems are often classified as "N-way speakers", where N indicates the number of separate frequency bands, usually separated by an electrical filter called a crossover. A 2-way system will have woofer and tweeter sections; a 3-way system a combination of woofer, tweeter, and mid-range speakers, and so on.

Contents [hide]
1 History
2 Driver design
2.1 Driver types
2.1.1 Full range drivers
2.1.2 Subwoofer
2.1.3 Crossover
2.2 Enclosures
2.2.1 Wiring connections
3 Specifications
4 Electrical characteristics of a dynamic loudspeaker
4.1 Electromechanical measurements
4.2 Efficiency vs. Sensitivity
5 Interaction with the listening environment
5.1 Loudspeaker placement
6 Loudspeaker directivity
6.1 Point sources
6.2 Line sources
7 Other driver designs
7.1 Horn loudspeakers
7.2 Piezoelectric speakers
7.3 Electrostatic loudspeakers
7.4 Heil air motion transducers
7.5 Plasma arc speakers
7.6 Digital speakers
8 References
9 See also
10 External links



[edit] History
Alexander Graham Bell patented the first loudspeaker as part of his telephone in 1876. This was soon followed by an improved version from Ernst Siemens in Germany and England (1878). Nikola Tesla is believed to have created a similar device in 1881[1]. The modern design of moving-coil drivers was established by Oliver Lodge in (1898)[2]. The moving coil principle was patented in 1924 by Chester W. Rice and Edward W. Kellogg.

These first loudspeakers used electromagnets because large, powerful permanent magnets were not available at reasonable cost. The coil of an electromagnet, called a field coil, was energized by direct current through a second pair of connections to the driver. This winding usually served a dual role, acting also as a choke coil filtering the power supply of the amplifier to which the loudspeaker was connected.

The quality of loudspeaker systems until the 1950s was, to modern ears, poor. Continuous developments in enclosure design and materials have led to significant audible improvements. The most notable improvements in modern speakers are improvements in cone materials, the introduction of higher temperature adhesives, improved permanent magnet materials, improved measurement techniques, computer aided design and finite element analysis.


[edit] Driver design

Cut-away view of a dynamic loudspeakerThe most common type of driver uses a lightweight diaphragm connected to a rigid basket, or frame, via flexible suspension which constrains a coil of fine wire to move axially through a cylindrical magnetic gap. When an electrical signal is applied to the voice coil, a magnetic field is created by the electric current in the coil which thus becomes an electromagnet. The coil and the driver's magnetic system interact, generating a mechanical force which causes the coil, and so the attached cone, to move back and forth and so reproduce sound under the control of the applied electrical signal coming from the amplifier. The following is a brief discussion of the individual components of this most common type of loudspeaker.

The diaphragm is usually manufactured in a cone or dome shaped profile. Numerous materials may be used, but the most common are paper, plastic and metal. The ideal material would be stiff, light and well damped. In practice, all three of these criteria cannot be met, and thus driver design involves tradeoffs. Paper is light and well damped, but not stiff. Metal can be made stiff and light, but it is not well damped. Plastic can be light, but typically the stiffer it is made, the less well-damped it is. As a result, many cones are made of some sort of composite. This can either be a sandwich construction or simply a coating to stiffen or damp a cone.

The basket or frame must be designed for rigidity to avoid deformation which could cause the voice coil to rub against the magnet structure. Baskets are typically cast or stamped metal, although molded plastic baskets are becoming common, especially for inexpensive drivers. The frame plays a secondary role in conducting heat away from the coil.

The suspension system keeps the coil centered in the gap and provides a restoring force to make the speaker cone return to a neutral position after moving. A typical suspension system consists of two parts: the "spider", which connects the diaphragm or voice coil to the frame and provides the majority of the restoring force; and the "surround", which helps center the coil and allows free movement. The spider is usually made of a corrugated fabric disk. The surround can be a roll of rubber or foam or a corrugated fabric, attached to the outer circumference of the cone and to the frame.

The voice coil wire is usually copper, though aluminum, or rarely silver, may be used. Voice coil wire can be round, rectangular, or hexagonal, giving varying amounts of wire volume coverage in the available magnetic gap. The coil is oriented coaxially inside the gap, a small circular volume (a hole, slot, or groove) in the magnetic structure within which it can move back and forth. The gap establishes a concentrated magnetic field between the two poles of a permanent magnet; the outside of the gap being one pole and the center post (a.k.a. pole-piece) being the other. The center post and back-plate are sometimes a single piece called the yoke.

Modern driver magnets are almost always permanent and made of ceramic, ferrite, Alnico, or, more recently, rare earth. The size and type of magnet and the magnetic circuit differ depending on design goals. A current trend in design, due to increases in transportation costs and a desire for smaller, lighter devices (as in many home theater multi-speaker installations), is the use of rare earth magnet instead of ferrite types.

Driver design, and the combination of one or more drivers into an enclosure to make a speaker system, is both an art and science. Adjusting a design to improve performance is done using magnetic and material science theory, high precision measurements, as well as experienced listeners. Designers can use an anechoic chamber to ensure the speaker can be measured independently of room effects, or any of several electronic techniques. Some developers eschew anechoic chambers in favor of specific standardized room set-ups intended to simulate real-life listening conditions. Some of the issues speaker designers must confront are lobing, phase effects, off axis response, crossover complications, and psychoacoustics.

Most loudspeaker drivers are currently manufactured in China. The fabrication of finished loudspeaker systems is segmented, depending largely on price point. High-end speaker systems are usually made in the same region as their target markets and can command prices of $10,000 per pair and up. The lowest-priced speaker systems are mostly manufactured in China or other low-cost manufacturing locations. Although the manufacture of drivers has become essentially commoditized, the fabrication and subsequent sale of finished speaker systems still carry high profit margins. Partly for this reason, manufacturers are increasingly combining power amplifier electronics (a typically lower profit item) with finished speaker systems to create "powered speakers" with an overall higher market value.


[edit] Driver types

Exploded view of a dome tweeterA woofer is a driver capable of reproducing low (bass) frequencies. The usable frequency range varies widely according to design. Some woofers can cover the audio band from lowest bass to 3 kHz, while others only work up to 1 kHz or less. Some woofers are capable of very deep bass performance in an enclosure that is large enough and properly braced. Others woofers become unusable or highly distorting below 50 or 60 Hz, and so listeners who want to listen to music with very deep bass may need a subwoofer (see below).

A tweeter is a driver capable of reproducing the higher end of the audio spectrum, usually from around 3-5 kHz up to 20 kHz and beyond.

A mid-range speaker, also called a squawker, is designed to cover the middle of the audio spectrum, typically from a few hundred Hertz to about 4-5 kHz. Midranges are used when the other drivers are incapable of adequately covering the full audio range without them. They also increase system maximum output, as tweeters in 3-way systems can be spared the difficult requirement to reproduce lower frequencies; this increases their maximum sound output before damage.


[edit] Full range drivers
A full-range driver is designed to have as wide a frequency response as possible. These drivers are small, typically 2 to 6 inches (5 to 16 cm) in diameter to permit reasonably high frequency response, but this means they often have limited low distortion sound output at low frequencies and limited power handling capacity (due to a small voice coil).

They often employ an additional cone called a whizzer, a small, light cone attached to the woofer's apex near the dust cap, to extend the high frequency response and broaden the high frequency directivity. The main cone is built so as to flex more in this region at high frequencies than the rest of the cone. The result is that the main cone delivers the low frequencies and the whizzer cone contributes most of the higher frequencies. Since the whizzer cone is smaller than the main diaphragm, dispersion at high frequencies is improved over a driver with a single larger diaphragm. Full range drivers are one approach to avoiding the possible audible effects of multiple driver systems caused by non-coincident driver location and crossover issues.


[edit] Subwoofer
A subwoofer is a woofer driver used only for the lowest part of the audio spectrum. A typical subwoofer only reproduces sounds below perhaps 120 Hz; some can go lower than 20 Hz. Because the intended range of frequencies is limited, subwoofer design is usually simpler, often consisting of a single, subwoofer enclosed in a suitable (often bass reflex) cabinet. To accurately reproduce very low bass notes without unwanted resonance, subwoofers have to be large enough and properly braced. Subwoofers are often supplied with power amplifiers and electronic filters, with additional controls relevant to low frequency reproduction, such as phase switches built directly into the cabinet. These subwoofers are known as "active subwoofers". Some subwoofer systems also include sophisticated systems utilizing accelerometers or back EMF sensors to sense cone movement. The actual motion of the cone is compared to the input signal many times per second and the feedback circuitry applies continuous correction to the drive signal to enable the woofer to reproduce the input signal with less distortion. These last are often called "servo" or "motional feedback" subwoofers.


[edit] Crossover
Main article: Audio_crossover
In a multiple driver (i.e. 2-way, 3-way, etc...) loudspeaker system, some means must be provided to separate the frequency band into sections so that each driver will produce the frequency range it is designed for, and to reduce the interference between the drivers. This separation of the frequencies is accomplished using a type of filter circuit called a crossover. The ideal crossover would have no overlap, but this is not achievable in practice with standard analog filters. The vast majority of loudspeakers use a passive crossover circuit. Passive crossover circuits use only capacitors, inductors, and resistors, which are known as passive components. Active crossovers use extra amplification stages to divide the frequency range before the signal is amplified. These require a separate amplifier for each frequency range. There are some inherent advantages to active crossovers, but the added expense and complexity makes them most prevalent in professional sound applications.


[edit] Enclosures
Main article: Loudspeaker enclosure

A 4-way speaker system. The cabinet is narrow to reduce a diffraction effect called the 'baffle step'.Most loudspeaker systems consist of drivers mounted in an enclosure, or cabinet. The main physical role of the enclosure is provide a place to mount the drivers. Perhaps the simplest enclosure is a baffle, just a flat board with the drivers mounted to it. This simple enclosure has the disadvantage that at frequencies with a wavelength longer than the baffle dimensions the antiphase radiation from the rear of the cone is free to interfere with the front radiation and will cause uneven response and a loss of bass. If the baffle is made infinitely large, this problem goes away.

Since infinite baffles are impractical, most enclosures function by containing the rear radiation from the cone. The simplest is a sealed box. The sealed enclosure prevents transmission of the sound emitted from the rear of the loudspeaker to the listening space by ideally being rigid and airtight. Techniques used to reduce transmission of sound through the walls of the cabinet include thicker cabinet walls, lossy wall material, internal bracing, curved cabinet walls or more rarely visco-elastic materials or thin lead sheeting applied to interior enclosure walls.

However, this rigid enclosure will then induce internal reflection of sound which can then be retransmitted through the loudspeaker cone; again resulting in degradation of sound quality. This is reduced through internal absorption through the use of absorptive materials (often called "damping") such as fiberglass, wool or synthetic fiber batting within the enclosure. The internal shape of the enclosure can be designed to reduce this by reflecting sounds away from the loudspeaker where they may then be absorbed.

Many other enclosure types exist which attempt to modify the rear radiation, which is half of the energy radiated by the driver, so that it may add constructively to the output from the front of the cone. Many designs which do this (Bass reflex, passive radiator, transmission line, etc...) are often used to extend the low frequency response of the speaker system.

In an attempt to make the transition between drivers as seamless as possible, system designers have also attempted in recent years to time-align or phase adjust the drivers, which often involves moving one or more drivers forward or back, so that the acoustic centers of the drivers is in the same vertical plane. This sometimes involves tilting the face of a floor-mounted speaker back, or providing separate enclosure mounting for the drivers, or, less commonly, using electronic techniques to achieve the same effect. These attempts account for some of the unusual cabinet arrangements in speaker systems.

Another issue designers must manage is sound wave diffraction caused by the surfaces (face plate, cabinet, etc.) in which a driver is mounted. This is usually a problem at higher frequencies, as those wavelengths are similar to, or smaller than, cabinet dimensions. The problem is addressed by rounding the front edges of the cabinet or by using a smaller or narrower enclosure, or by strategic arrangement of the drivers. Sometimes, an absorptive layer such as felt is added to the mounting surface around a driver to reduce such effects.


[edit] Wiring connections

Five-way binding posts on a loudspeaker connected using banana plugs.Most loudspeakers use two wiring points to connect to the source of the signal (for example, to the audio amplifier or receiver). This is usually done using binding posts, or spring clips on the back of the enclosure. If the wires for left and right speakers (in a stereo setup) are not connected in phase with each other (the + and - connections on the speaker and amplifier should be connected to each other) the loudspeakers will be out of phase and destructive sound wave interference will occur when a common signal is sent to each speaker. In this case, any motion one cone (usually the woofer) makes will be opposite to the other. This type of wiring error doesn't damage speakers but does create inverse sound waves that partially cancel those from the other speaker. Due to the spacing of the speakers, the bass frequencies are where this phenomenon is most apparent.


[edit] Specifications

Specifications label on a loudspeakerSpeaker specifications generally include:

Speaker or driver type (individual units only) – Full-range, woofer, tweeter or mid-range.
Rated Power – Nominal or continuous power and peak or maximum short-term power that the loudspeaker can handle (that is, maximum allowed input power without thermally destroying the loudspeaker. It is not the power that the passive loudspeaker produces). Under special conditions, such as overdriving at very low frequencies or via sine wave input at higher frequencies, a loudspeaker may be damaged at much less than rated power.
Impedance – typically 4 Ω (ohms), 8 Ω, etc.
Baffle or enclosure type (enclosed systems only) – Sealed, bass reflex, etc.
Number of drivers (complete speaker systems only) – 2-way, 3-way, etc.
and optionally:

Crossover frequency(ies) (complete multi-driver systems only) – The frequency or frequencies where electrical filtering occurs.
Frequency response – The measured or specified variance in sound pressure level to a constant input over a specified range of frequencies, often including a variance such as within +/- 2.5 dB.
Thiele/Small parameters (individual drivers only) – these include the driver's Fs (resonance frequency), Qts (the driver's Q or damping factor at resonance), Vas (the equivalent air compliance volume of the driver), etc.
Sensitivity – The sound pressure level produced by a loudspeaker, usually specified in dB, measured at 1 meter with an input of 1 Watt or 2.83 Volts. This rating is often inflated by manufacturers.

[edit] Electrical characteristics of a dynamic loudspeaker
Main article: Electrical characteristics of a dynamic loudspeaker
The load a driver presents to an amplifier consists of a complex electrical impedance, a combination of resistance, and both capacitive and inductive reactance, reflecting the properties of the driver, its mechanical motion, the effects of crossover componenents (if any), and the effects of air loading on the driver as modified by the enclosure. Most amplifiers (amps) output specifications are given at a specific power into an ideal resistive load. However, a loudspeaker with a nominal impedance of 8 Ω does not really have a constant resistance. Instead, the voice coil is inductive, the enclosure changes the characteristics of the driver, and a passive crossover between the drivers and the amplifier contributes its own variations.


[edit] Electromechanical measurements
Fully characterizing the sound output of a loudspeaker in detail is difficult (for example, phase characteristics vs. frequency, impulse response at various frequencies, directivity vs. frequency, distortion vs. SPL output (eg, harmonic, intermodulation vs SPL output, compression, etc), stored energy (that is, ringing) vs. frequency and output level, small signal vs. large signal performance, etc.), but the raw sound pressure level output is rather easier to measure. The sound pressure level (SPL) a loudspeaker produces is measured in decibels (dBspl).


[edit] Efficiency vs. Sensitivity
Loudspeaker efficiency is defined as the sound power output divided by the electrical power input. Most loudspeakers are actually very inefficient transducers. Only about 1% of the electrical energy sent by an amplifier to a typical home loudspeaker is converted to the acoustic energy we can hear -- the remainder is converted to heat, typically in the voice coil and magnet assembly. The main reason for this is the difficulty of achieving proper impedance matching between the acoustic impedance of the drive unit and that of the air into which it is radiating.

Driver ratings based on the SPL for a given input voltage (corresponding closely to power input for a particular driver impedance) are known as sensitivity ratings and are, approximately, equivalent to efficiency. Sensitivity is usually defined as so many dB at 1 W electrical input, measured at 1 meter. The voltage used is often 2.83 VRMS, which happens to be 1 watt into an 8 Ω (nominal) speaker impedance (nominally true for many speaker systems). Measurements taken with this reference are quoted as dB with 2.83 V @ 1 m.

The sound pressure is measured at (or scaled to be equivalent to a measurement taken at) one meter from the loudspeaker and on-axis or directly in front of it under the conditions that the loudspeaker is radiating into an infinitely large space and mounted on an infinite baffle. Clearly then, sensitivity does not correlate precisely with efficiency as it also depends on the directivity of the driver being tested and the acoustic environment in front of the actually deployed loudspeaker. As a simple example, a cheerleader's horn makes more sound output in the direction it is pointed than the cheerleader could by herself, but the horn did not improve or increase the cheerleader's total sound power output much, it just focused it into a smaller space.

Typical home loudspeakers have sensitivities of about 85 to 95 dB for 1 W @ 1 m - an efficiency of 0.5-4%.
Sound reinforcement and public address loudspeakers have sensitivities of perhaps 95 to 102 dB for 1 W @ 1 m - an efficiency of 4-10%.
Rock concert, stadium PA, marine hailing, etc speakers all have higher sensitivities -- maybe 103 to 110 dB for 1 W @ 1 m - an efficiency of 10-20%.
A driver with a higher maximum power rating cannot necessarily be driven to louder levels than a lower rated one, since sensitivity and power handling are independent. In the examples which follow, assume for simplicity that the drivers being compared have the same electrical impedance, are operated at the same frequency which is within both driver's respective pass bands, and that power compression is and distortion are low. For the first example, a speaker 3 dB more sensitive than another will produce double the sound pressure level (or be 3 dB louder) for the same power input. Thus a 100 W driver ("A") rated at 92 dB for 1 W @ 1 m sensitivity will output twice as much acoustic power as a 200 W driver ("B") rated at 89 dB for 1 W @ 1 m when both are driven with 100 W of input power. For this particular example, when driven at 100 W, speaker A will produce the same SPL, or loudness, speaker B would produce with 200 W input. Thus a 3 dB increase in sensitivity of the speaker means that it will need half the amplifier power to achieve a given SPL; this translates into a smaller, less complex power amplifier and, often, to reduced overall cost.

It is not possible to combine high efficiency, especially at low frequencies, with compact enclosure size, and adequate low frequency response. One can, more or less, only choose two of the three parameters when designing a speaker system. So, for example, if extended low frequency performance and a small box size are important, one must accept low efficiency. This rule of thumb is sometimes called Hoffman's Iron Law (after J. A. Hoffman, the H in KLH).


[edit] Interaction with the listening environment
The interaction of a loudspeaker system with its environment is complex and is largely out of the designer's control. Most listening rooms present a more or less reflective environment, depending on size, shape, volume, and furnishings. This means the sound reaching a listener's ears consists not only of direct sound, but also of that sound delayed by traveling to and from (and being modified by) reflections from one or more surfaces. These reflected sound waves, when added to the direct sound, cause cancellation and addition at certain frequencies, changing the timbre and character of the signal being reproduced. Our brains are very sensitive to these small variations. This is part of the reason why a loudspeaker system sounds different at different listening positions or in different rooms.

A significant factor in the sound of a loudspeaker system is the amount of absorption and diffusion present in the environment. Clapping one's hands in an empty room, without draperies or carpet, will produce a zippy fluttery echo which is due both to a lack of absorption and to reverberation (that is, repeated echoes). The addition of hard surfaced furniture, wall hangings, and shelving will change the echoes, due primarily to the diffusion caused by somewhat reflective objects with shapes and textures having sizes on the order of the sound wavelengths being diffused. This somewhat breaks up the simple reflections otherwise caused by flat walls, floors and ceilings, and spreads the reflected energy of an incident wave over a larger angle on reflection.

Adding carpet, curtains, tapestries, people, or soft surfaced furniture will further change the interaction of a loudspeaker with the room by absorbing sound at various frequencies and reducing reflections at those frequencies. By and large, the thinner a material is, the less likely it will have an effect at low frequencies. An overabundance of absorption at high frequencies can be caused by large areas of absorptive materials and can cause a speaker system to sound deficient at higher frequencies, and likewise minimal absorption can cause an otherwise adequate loudspeaker to sound too bright or sibilant at those frequencies.


[edit] Loudspeaker placement
For good sound in a home environment, a listening room should have a balance of diffusion and absorption. Most systems will sound best when the speakers are set up more or less symmetrically with respect to the listener and also to room boundaries. Early reflections (the first reflection of a particular sound) do the most to color the sound (due to the so-called Haas effect in psychoacoustics), so placing speakers too near the rear or side walls is generally something to be avoided, although judicious use of absorbing or diffusing materials can somewhat moderate an otherwise poor placement location. Mounting a speaker in a wall (or in a bookshelf with books flush with the baffle) somewhat removes diffractive boundary concerns, but limits placement flexibility. In professional applications, placement is largely controlled by the location of the listening audience, required appearance (for example, prominence or invisibility), and available space. Fine adjustment is often not possible.

Another factor in room acoustics is a phenomena called standing waves. A one dimensional example is sound bouncing between two reflective boundaries. Sound resonates, or repeatedly reflects at particular frequencies, if the distance between the boundaries corresponds to an integral number of half wavelengths. Since sound travels at ~345 m/s, a pair of reflective boundaries separated by 5 meters will cause resonances at 34.5 Hz, 69 Hz, 103.5 Hz ..., recalling that wavelength is the speed of sound divided by the frequency. It is best, if possible, to arrange that no room wall length or height is simply related to any other. A cubical listening room would be most resonant since all dimensions are identical, with walls, floor and ceiling parallel, thus reinforcing the resonance modes. One approach is to ensure that each room dimension is related to another by the Golden Mean, which will ensure that the unavoidable reflections between walls are not reinforced by any others.

In a typical rectangular listening room, this resonant phenomenon happens in three dimensions, and there are even more complex interactions that involve four or even all six boundary surfaces. It is primarily an issue for low frequencies which are not much affected by such things as furniture or its placement. In addition, the location of the loudspeakers, and the listener, with respect to room boundaries affect how strongly the resonances are excited. Many people are familiar with certain locations in a room, club, or building which have much more, or less, bass - most usually near room walls or corners. This is because standing wave patterns are most pronounced in these locations and at lower frequencies, below the Schroeder frequency - typically around 200-300 Hz, depending on room size.


[edit] Loudspeaker directivity
This is an important issue because it affects the frequency balance of sound a listener hears, and also the interaction of the speaker system with the room.

In general, a sound source will radiate of one of four basic ways: as a point source, a line source, a planar source or a 3D source.

An extremely small point source is often considered ideal, because it radiates all frequencies equally in all directions in a spherical radiation pattern, and thus favors none. A theoretical line sources may be finite or effectively infinite, and will radiate sound in a cylindrical pattern. These first two source types are not actually practical, although real sound sources may approximate them, especially at some frequencies. In real life, most speaker systems and individual drivers are actually complex 3D shapes such as cones and domes.

Some drivers, and some enclosures (e.g. horns) take advantage of directivity in that, rather than radiating in a wide pattern, they focus sound into a constrained pattern. This is desirable for large areas such as theaters, concert halls, arenas and outdoor areas where the listener(s) may be a great distance from the sound source and yet should still hear well.

Less well understood are the psychoacoustic consequences of particular kinds of directivity. For instance, Amar Bose delivered a paper in 1968 to an Audio Engineering Society conference entitled "On the Design, Measurement and Evaluation of Loudspeakers" (reprints are available from the AES: http://www.aes.org/e-lib/browse.cfm?elib=1390), in which he discussed the issue of reflected versus direct sound, and that in live performance most, if not all, listeners are located in the sound field for which reflected sound dominates. Only a tiny fraction of the sound reaching the pinna of the ear arrives as direct sound from the musical instruments; in a typical home listening environment, speaker systems at that time delivered the higher frequencies directly at the ears of the listener. Does directionality matter in this context (i.e. reproduction of recorded sound)? Physics and engineering cannot resolve this issue; it lies in the domain of psychoacoustic and recording practice. The experiments Amar Bose conducted at MIT in the 1960s, reported in this paper, convinced him that a dominance of reverberant sound is important to the perception of quality in sound reproduction, IF one's standard of quality is informed by the experience of live performances.


[edit] Point sources
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Please see the discussion on the talk page.

Point sources can be approximated (at least at low frequencies) as a planar form that creates a sound wave which becomes more directional as frequency increases, because the wavelength of the sound wave becomes small compared to the size of the diaphragm. That is, the intensity of the sound produced varies depending on the listener's angle relative to the central axis of the speaker.

A common variation on the dynamic loudspeaker cone design uses a dome as the moving part instead of the familiar inverted cone. Contrary to intuition, making the moving surface a dome rather than an inverted cone does not always help to direct sound evenly in a half-spherical space. The dome is used primarily because, in the case of a tweeter, its radiating surface is smaller than the voice coil and because a conical shape is difficult to fit in the tweeter structure -- unless specially modified, the magnet system's pole piece will mechanically interfere with cone motion at high amplitudes. Some tweeters do have inverted domes, however, but the pole piece is specially configured to accommodate the dome's shape. Some tweeters (TDL and other manufacturer use this shape) use bullet-shaped domes instead of domes with a constant radius. The intent is to reduce or eliminate bending in the center of the dome (consider that an egg shape is harder to push in at the pointed end than at the other). Finally, some manufacturers leave out the center of the dome altogether and only use the outer ring (called a ring radiator) to altogether avoid distortions of the inner part of the dome due to bending effects (for example, some models from Scan-Speak, Kea Audio, Vifa etc). A ring radiator also has better directivity (that is, is less directive) than a dome. A typical one inch dome tweeter begins to be directive at about 8000 Hz, below this frequency it approximates a point source, above this frequency, it becomes increasingly directional. At distances more than ~7 times the diameter of the cone or dome, the response is essentially that of a flat plane, at closer distances, the exact shape of the diaphragm becomes increasingly important.

Two dimensional and three dimensional sound sources can be monopolar, dipolar or bipolar. Most planar (that is, flat diaphragm) drivers are dipolar, which means that sound from the rear of the diaphragm is permitted to freely radiate. When the rear radiation is absorbed or trapped in a box, the diaphragm becomes a monopole radiator. Bipolar speakers, made by mounting in-phase monopoles on opposite sides of a box, are a method of approximating a point source or pulsating sphere.

Various manufacturers use assorted driver mounting arrangements, and the resulting radiation patterns, to more closely simulate the way sound is produced by real instruments, or to mimic one of the ideal sound source types, or simply to create a controlled energy distribution. Most professional audio speaker systems use horns or other dispersion control techniques because broad dispersion is a liability in many commercial situations such as concert sound or public address contexts.

The Manger bending wave transducer uses a bending wave scheme, in which vibration waves start from the center of a round flat diaphragm and travel to the outside. The rigidity of the material increases from the center to the outside. Short wavelength sound therefore radiate primarily from the inner area, while longer waves reach the edge of the speaker. To prevent reflections, long waves are absorbed by a surrounding damper. The Manger transducer covers the frequency range from 80 Hz to 35,000 Hz, and is close to an ideal point sound source. The Walsh loudspeaker systems from Ohm Acoustics have been quite similar in their bending scheme, though different in numerous details.

Coaxial speakers have been made commercially since the 1930s. These approximate a point source by moving the radiating axes of the various drivers close to the same point, usually with benefits in polar response. Coaxial mounting eliminates crossover lobing (that is, interference between drivers caused by non coincident placement). The woofer cone often acts as a horn in many respects. The technique of using concentric radiating elements for a multiway system has been used by several manufacturers, notably Technics. Cabasse recently published a paper analyzing 3-way and even 4-way coaxial speakers using concentric ring-shaped radiators. Several manufacturers (for example, Tannoy, Eminence, etc.) still build 2-way coaxial drivers in which the tweeter fires through a horn that passes through the woofer pole piece, and several (for example, KEF, SEAS, Kea-Audio, Tannoy etc.) build coaxial units in which the tweeter is mounted on the woofer pole piece. The small form factor this last approach requires has been made more effective by recent developments in rare earth magnets.[3]

Several manufacturers have attempted to simulate a point source by approximating a pulsating sphere. In the 1960s, Amar Bose (an MIT Professor) designed a one-eighth sphere loudspeaker system covered in small full-range drivers for room corner placement. The 1801 produced a wavefront very like that of an ideal sphere when wall reflections were included. Few were built and the system was not a commercial success, but it gave rise to commercially successful speaker system designs (the 901, most importantly) which also use multiple small drivers pointed in various directions to create a mixture of direct and reflected sound claimed to approximate that of a concert hall. In the 1801 and the 901, the small drivers involved were not actually inherently full-range and required considerable equalization to provide adequate low frequency performance and to compensate for decreasing high frequency performance. Especially at low frequencies, this approach demanded rather more amplifier power than competing speakers of the time. Both techniques have remained somewhat controversial.

The Ohm speaker drivers, whose principle was invented by Lincoln Walsh, use a single voice coil/cone mounted vertically, firing downwards into the top of the cabinet, but instead of the normal almost flat cone, has an extended cone entirely exposed at the top of the speaker. The usual problem with designing a cone driver is how to keep the cone as stiff as possible (without adding too much mass) so that it moves as a unit, and does not support traveling waves nor distort during cone breakup. The Walsh driver was so designed that the entire purpose of the cone's motion was to generate traveling waves down the cone from the magnetic motor (that is, voice coil and magnet structure) at the top. As the waves moved down the cone, the effect was to reproduce a 360 degree wavefront at all frequencies, more or less like a cylinder. This created a very effective omni-directional radiator (although it suffered the same "planarity" effect as ribbon tweeters for higher-frequency sounds ) and eliminated all problems of multiple drivers, such as crossover issues, phase anomalies between drivers, etc. However, in practice it was found necessary to use a very complex and expensive cone made of various materials along its length.

High fidelity speaker systems of this design are still being produced by Ohm in the US, and in Germany, by German-Physik and, as a variant, by Manger. This approach has not been used in professional sound reinforcement, most likely due to the delicacy of the physically large cone structure and the inherent cylindrical directivity.


[edit] Line sources
A ribbon speaker consists of a thin metal-film ribbon suspended in a magnetic field. The electrical signal is applied to the ribbon which moves with it, thus creating the sound. The advantage of a ribbon driver is that the ribbon has very little mass; thus, it can accelerate very quickly, yielding very good high-frequency response. Ribbon loudspeakers are often very fragile -- some can be torn by a strong puff of air. Most ribbon tweeters emit sound in a dipole pattern; a very few have backings which limit the dipole radiation pattern. Above and below the ends of the more or less rectangular ribbon, there is less audible output due to phase cancellation, but the precise amount of directivity depends on ribbon length. Ribbon designs generally require exceptionally powerful magnets which make them costly to manufacture. Ribbons have a very low resistance that most amplifiers cannot drive directly. A step down transformer is therefore typically used to increase the current through the ribbon. The amplifier "sees" a load that is the ribbon's resistance times the transformer turns ratio squared. The transformer must be carefully designed so that its frequency response and parasitic losses do not degrade the sound, further increasing cost and complication relative to conventional designs.

Planar magnetic speakers (having printed or embedded conductors on a flat diaphragm) are sometimes described as "ribbons", but are not truly ribbon speakers. The term planar is generally reserved for speakers which have roughly rectangular shaped flat radiating surfaces. Planar magnetic speakers consist of a flexible membrane with a voice coil printed or mounted on them. The current flowing through the coil interacts with the magnetic field of carefully placed magnets on either side of the diaphragm, causing the membrane to vibrate more or less uniformly and without much bending or wrinkling. The driving force covers a large percentage of the membrane surface and reduces resonance problems inherent in coil-driven flat diaphragms. Many designs touted as "ribbons" are in fact planar magnetic. Many of these designs have small cavities between the magnet structures and the diaphragm. This is not ideal and it sometimes creates a "cavity resonance" response peak that requires corrective filtration. Failure to correct this cavity resonance is a cause of the steely or shrill sound sometimes attributed to these designs.

There have also been many attempts to reduce the size of speaker systems, or alternatively to make them less obvious. One such attempt was the development of voice coil driven 'exciters' mounted to flat panels to act as sound sources. These can then be made in a neutral color and hung on walls where they will be less noticeable than many speakers, or can be deliberately painted with patterns in which case they can function decoratively. An example is Wharfedale Pro's 'Loudpanel' series. There are two related problems with flat panel techniques: first, a flat panel is necessarily more flexible than a cone shape in the same material, and therefore will move as a single unit even less, and second, resonances in the panel are difficult to control, leading to considerable distortions. Some progress has been made using such lightweight, rigid, yet damped, materials as Styrofoam, and there have been several flat panel systems commercially produced in recent years.

A newer implementation of the flat panel speaker system involves an intentionally flexible panel and an "exciter", mounted off-center in a location such that it excites the panel to vibrate, but with minimal resonances. Speakers using NXT techniques design methods can reproduce sound with a wide directivity pattern (paradoxically somewhat like a point source) and have been used in some computer speaker designs and a few small 'shelf systems' from such manufacturers as TEAC and Philips.


[edit] Other driver designs
Other types of drivers which depart from the most commonly used electro-dynamic driver mounted in an enclosure include:


[edit] Horn loudspeakers

Painting of Nipper, used by Gramophone Ltd and Victor Talking Machine, UK, & then RCA, USHorn speakers have been designed and built since the late 19th century; one is prominent in the RCA logo with Nipper the dog listening to His Master's Voice. Horns using modern electrodynamic drivers are a more recent development beginning shortly after the First World War. The increasing cross-sectional area of the horn allows for a greater mechanical advantage of the driver against the resistance of air, increasing the efficiency of the driving element. An efficient home loudspeaker system has a sensitivity of around 90 dB @ 2.83 volts (1 watt @ 8 Ohms) @ 1 Meter distance, while several home-use horn loaded speakers are rated as high as 100 dB @ 2.83 volts (1 watt @ 8 Ohms) @ 1 Meter. This is a tenfold increase in output at one watt, resulting in an output level which would require 10 watts from the speaker rated at 90 dB sensitivity, and is invaluable in some applications. The length and cross sectional mouth area required to create a bass or sub-bass horn may necessitate a horn several feet long. Due to the large volume that such a horn occupies, it is often necessary to fold the horn in order to allow it to fit its environment. Formerly, largely after WWII and before the stereo era, horns whose mouths took up much of a room wall were not uncommon amongst hi-fi fans. Such installations became much less acceptable when two were required, and entirely unthinkable in modern, multi-channel home systems.

Few full-range horns are being commercially produced for home use, and those which are have very high prices. But there is an active DIY horn building community around the world which has produced some visually striking enclosures, some claimed to be audibly excellent as well. More common are 'short horns' (of a practical, though still large, size) used for professional sound work. These are typically bass reflex enclosures usually with two large drivers (12" or 15") firing into a common horn with a very large throat. The horn in these cases is more used for dispersion control than acoustic loading at low frequencies. The Altec Lansing Voice of the Theater model is an example, first used in movie theater sound five decades ago.


[edit] Piezoelectric speakers
Piezoelectric speakers are frequently used as beepers in watches and other electronic devices, and are sometimes used as tweeters in less-expensive speaker systems, such as computer speakers and portable radios. Piezoelectric speakers have several advantages over conventional loudspeakers:

they have no voice-coil, therefore there is no electrical inductance to manage
it is easy to couple high-frequency electrical energy into the piezoelectric transducer since the transducers are resistant to overloads which would normally destroy the voice coil of a conventional loudspeaker.
they are an inherently capacitive electrical load so they usually do not require an external cross-over network. They can simply be placed in parallel with conventional inductive voice coil drivers.
There are also disadvantages:

their frequency response, in most cases, is inferior to that of other technologies. This is why they are generally used in single frequency (beeper) or non-critical applications
some amplifiers cannot drive capacitive loads well; this can cause high frequency oscillation, which results in distortion or damage to the amplifier.

[edit] Electrostatic loudspeakers
Electrostatic loudspeakers give a more linear response than electromechanical voice coils, though for considerably reduced maximum motion amplitude. The entire diaphragm is driven by electrostatic charges and is very closely controlled. For many years electrostatic loudspeakers had a reputation as an unreliable and occasionally dangerous product. A primary disadvantage was that the signal must be converted to a very high voltage at low current, which was problematic for reliability and maintenance. High voltage charges attract dust, and many of these the speakers developed a tendency to arc, particularly where the dust provided a partial discharge path. Arthur Janszen was granted U.S. Patent 2,631,196 in 1953 for the practical electrostatic design.

Full range electrostatic loudspeakers are large by nature. An early model, the KLH Model 9, was taller than most people. In addition, electrostatics are inherently dipole radiators and cannot, in practice, be used in enclosures to increase their efficiency as with common cone drives. In electrostatic loudspeakers, diaphragm excursion is limited to fractions of a millimeter whereas more ordinary dynamic cone loudspeakers can usually move many millimeters, even up to centimeters in some instances. This means that the membrane of a full range electrostatic loudspeaker must be larger than an equivalent dynamic loudspeaker to produce even marginally acceptable low frequency performance and output level. Electrostatic tweeters have proven to be more practical and are more widely used.


[edit] Heil air motion transducers
Dr Oscar Heil invented this design in the 1960s. ESS, a California manufacturer, licensed it, employed Dr Heil, and produced a range of speaker systems using them as tweeters during the 1970s and 1980s. Radio Shack, a large US retail store chain, also sold speaker systems using them as tweeters for a time.

In this approach, a pleated diaphragm is mounted in a magnetic field and forced to close and open under control of a music signal. Air is forced from between the pleats in accordance with the imposed signal, generating sound. The drivers are less fragile than ribbons and considerably more efficient (and able to produce higher absolute output levels) than ribbon, electrostatic, or planar magnetic tweeter designs. At present, there are two manufacturers of these drivers, both in Germany, one of which produces a range of high end professional speakers using tweeters and midrange drivers based on the technology.


[edit] Plasma arc speakers
Plasma arc loudspeakers use electrical plasma as a driver. Since plasma has minimal mass, but is charged and therefore can be manipulated by an electric field, the result is a very linear output at frequencies far higher than the audible range. Problems of maintenance and reliability for this approach tend to make it unsuitable for mass market use. In 1978 Dr. Alan Hill of the Los Alamos National Laboratory designed the Hill Plasmatronics, an $8000 monster whose plasma was generated from compressed helium gas.[4] This avoided the ozone and nitrous oxide produced by RF decomposition of air in an earlier generation of plasma tweeters made by the pioneering DuKane Corporation, who produced the Ionovac (marketed as the Ionofane in the UK) during the 1950s. Currently, there remain a few manufacturers, all in Germany it seems, and a do it yourself design has been published.

A less expensive variation on this theme is the use of a flame for the driver, as flames contain ionized (electrically charged) gases.[5]


[edit] Digital speakers
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Digital speakers are an experimental but venerable technology, having been the subject of experiments by Bell Labs as far back as the 1920s. The design is simple; each bit drives a tiny speaker driver. A value of "1" causes that driver to be driven to full amplitude; a value of "0" causes it to be completely shut off. Increasingly significant bits drive speakers of twice the area of the previous (often in a ring around the previous driver).

The name is sometimes confused with speakers used with digital equipment, such as computer sound systems; they are not actually digital in this sense, being examples of typical if usually small speaker designs.

There are two problems with this design which has led to it being abandoned as impractical for the present. For a reasonable number of bits (required for adequate sound reproduction quality), the size of the system becomes very large. For example, a 16 bit signal compatible with the 16 bit audio CD standard, starting with a 2 square inch (13 cm²) driver for the least significant bit, would require a total area for the drivers of over 900 square feet (85 m²). Secondly, due to analog digital conversion, the effect of aliasing is unavoidable, so that the audio output is "reflected" at equal amplitude in the frequency domain, on the other side of the sampling frequency. Even accounting for the vastly lower efficiency of speaker drivers at such high frequencies, the result generates an unacceptably high level of ultrasonics accompanying the desired output.